Yes i mean add it to the dial string inside the {}
it only will work if the channel with the variable set is tied to the FS session obj.
sofia_reg.c 1122 is where it all happens
so if session is null there the var code won't work.
you can add some debug code there and try to figure out what's wrong.
I tried the following in conf/dialplan/extensions/7_inbound.xml:<action application="export" data="sip_use_gateway=broadview"/>
<extension name="broadview_inbound_9325">
<condition field="destination_number" expression="^12675379325|2675379325$">Also tried the following in conf/dialplan/public.xml:
<action application="transfer" data="1001"/>
</condition>
</extension>
<action application="export" data="sip_use_gateway=broadview"/>
<extension name="public_did_broadview">
<condition field="destination_number" expression="^(12675379324|2675379324|12675379325|2675379325)$">Neither helped. When you say add it to the dial string directly that calls it, I'm not sure what you mean (I know the general format of {var_name=var_value}, so that's not my question). Do you mean add it in front of the 1001 as the target of the transfer?
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
By the way, hangup DOES work properly if I create another gateway and name it 64.115.128.6. However, I'd love to get it working without having to create a duplicate gateway with a non-intuitive name. It's definitely a lot better than nothing to do it that way, but I'd prefer to have it work with the sip_use_gateway scheme you mention. I'm assuming I'm just doing something wrong with how sip_use_gateway should be specified in the XML configuration files. Can you tell what I am doing wrong?
On Fri, 31 Oct 2008, Anthony Minessale wrote:
Date: Fri, 31 Oct 2008 09:49:18 -0500
From: Anthony Minessale <anthony.minessale@gmail.com>
Reply-To: freeswitch-users@lists.freeswitch.org
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication
try using "export" instead of "set" or add it to the dial string directly that calls it
{sip_use_gateway=broadview}sofia/.......
On Fri, Oct 31, 2008 at 9:42 AM, Wellie Chao <wchao@yahoo.com> wrote:
Where do you recommend I put the sip_use_gateway=broadview action?
I have tried in the conf/dialplan/public.xml like so:
<extension name="public_did_broadview">
<condition field="destination_number" expression="^(12675379324|2675379324|12675379325|2675379325)$">
<action application="set" data="sip_use_gateway=broadview"/>
<action application="transfer" data="$1 XML default"/>
</condition>
</extension>
I've also tried in conf/dialplan/extensions/7_inbound.xml (a file I created that is pulled in via an include
pre-processor directive):
<extension name="broadview_inbound_9325">
<condition field="destination_number" expression="^12675379325|2675379325$">
<action application="set" data="sip_use_gateway=broadview"/>
<action application="transfer" data="1001"/>
</condition>
</extension>
I have a gateway named broadview in conf/sip_profiles/external. In both cases, I still get the following error on
the Freeswitch console:
2008-10-31 10:37:28 [ERR] sofia_reg.c:1089 sofia_reg_handle_sip_r_challenge() No Matching gateway found
On Fri, 31 Oct 2008, Anthony Minessale wrote:
Date: Fri, 31 Oct 2008 08:04:23 -0500
From: Anthony Minessale <anthony.minessale@gmail.com>
Reply-To: freeswitch-users@lists.freeswitch.org
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Hangup problem/SIP BYE lacking authentication
See what they said in the challenge?
WWW-Authenticate: Digest
realm="SipLocal",nonce="3e952db60fb8",stale=false,algorithm=MD5,qop="auth"
Since this is a spontaneous challenge (which i think is somewhat silly since it lets you talk on the phone for 40
minutes then makes you authenticate to hangup but *shrug*) FS does not know which gateway to use for credentials.
The realm they sent was SipLocal so FS is looking in its configuration for a gateway with that name.
The 2nd thing it tries is the host from the To: header (64.115.128.6). if there was a gateway with either of those
names,
it would find it.
So try naming your gateway SipLocal or 64.115.128.6
or you can try setting the variable sip_use_gateway=<whatever> on the channel which can give it a hint which
gateway to use.
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